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gerrit-public.fairphone.software
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fp2-dev
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platform
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chromium_org
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third_party
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webrtc
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7deb3351eab29ad7a89db0237a9e3ebbc04b1058
7deb335
Make FrameGeneratorCapturer own frame_generator.
by pbos@webrtc.org
· 11 years ago
eb7b0c4
Merging video_full_stack_tests and video_engine_tests.
by phoglund@webrtc.org
· 11 years ago
25b57c0
iOS: unbreak the build following r4546
by fischman@webrtc.org
· 11 years ago
67acd69
VideoSendStream SSRC test.
by pbos@webrtc.org
· 11 years ago
a4944f2
Lock resources in event_posix.cc.
by pbos@webrtc.org
· 11 years ago
96ff6ab
Added missing static_cast conversion.
by pbos@webrtc.org
· 11 years ago
8ce445e
Implementation and testing of PLI in new API.
by pbos@webrtc.org
· 11 years ago
49bc1b8
Fixes to padding when driven by encoder.
by stefan@webrtc.org
· 11 years ago
3207eaa
Made all integration tests use consistent naming.
by phoglund@webrtc.org
· 11 years ago
662ded4
Implementing APIs to set maximum and minimum for latency.
by turaj@webrtc.org
· 11 years ago
ece3d35
Added choice of decode error mode to loopback test.
by agalusza@google.com
· 11 years ago
7fc75bb
Update talk to 50918584.
by wu@webrtc.org
· 11 years ago
1e817c3
Roll chromium_revision 214260:217707 and gflags 45:84
by fischman@webrtc.org
· 11 years ago
e807da9
Fix OSX keydown detection. I noticed that the OSX implementation differs from Linux and Windows, and it will trigger continuously for a key that is pressed down. It would totally make sense to change this to a callback driven model, but that's a bigger change.
by niklas.enbom@webrtc.org
· 11 years ago
f594a6b
OpenSl bug: not matching playout and record sample rate led to high or low pitch audio (depending on if playout rate was higher or lower than record rate).
by henrike@webrtc.org
· 11 years ago
e155918
Revert 4547 "Isolate GYP target and .isolate files for tests"
by kjellander@webrtc.org
· 11 years ago
298bbdb
Isolate GYP target and .isolate files for tests
by kjellander@webrtc.org
· 11 years ago
fd6d89f
The video capture module for iOS.
by sjlee@webrtc.org
· 11 years ago
e416ab2
Remove ViEBase::Init() call from VideoCall.
by pbos@webrtc.org
· 11 years ago
c2014fd
Remove VideoEngine class from new VideoEngine API.
by pbos@webrtc.org
· 11 years ago
d171544
Disable CanTransmitExtraRtpPacketsWithoutError on Windows.
by pbos@webrtc.org
· 11 years ago
eca72bf
Revert r4539 "Disable racy part of RunsRtpRtcpTestWithoutErrors".
by marpan@webrtc.org
· 11 years ago
48bcf6f
Disable racy part of RunsRtpRtcpTestWithoutErrors.
by pbos@webrtc.org
· 11 years ago
c5e70b0
Add native_handle.h to gyp.
by wuchengli@chromium.org
· 11 years ago
73acde2
To allow the propagation of under-run in NetEq.
by minyue@webrtc.org
· 11 years ago
52c5c70
Replace MapWrapper with std::map<>.
by pbos@webrtc.org
· 11 years ago
8c8c87f
Updated WebRTC version to 3.39
by elham@webrtc.org
· 11 years ago
823a888
Signal when shutting down DirectTransport.
by pbos@webrtc.org
· 11 years ago
d893b3f
Avoid acquiring VCM::_receiveCritSect during decode callback.
by wuchengli@chromium.org
· 11 years ago
fe881f6
Run loopback tests with network thread.
by pbos@webrtc.org
· 11 years ago
bb6151d
Added Opus stereo support
by minyue@webrtc.org
· 11 years ago
f15cc82
Fix crash in screen capturer on Mac
by sergeyu@chromium.org
· 11 years ago
7d82c9d
Hand over loopback packets to a network thread.
by pbos@webrtc.org
· 11 years ago
4870c02
Don't pace out packets or generate padding when the pacer is disabled.
by stefan@webrtc.org
· 11 years ago
705b38d
Remove include_dirs from test/test.gyp.
by pbos@webrtc.org
· 11 years ago
03931c6
Remove unused unreferenced code in webrtc/
by pbos@webrtc.org
· 11 years ago
f43029b
Revert "Avoid acquiring VCM::_receiveCritSect during decode callback."
by wuchengli@chromium.org
· 11 years ago
b0af417
Avoid acquiring VCM::_receiveCritSect during decode callback.
by wuchengli@chromium.org
· 11 years ago
e915174
Allowing decoding with errors, when disabling nack.
by mikhal@webrtc.org
· 11 years ago
a6b178f
Fix duplicate code
by niklas.enbom@webrtc.org
· 11 years ago
a4a1afa
Delete Channels without ChannelManager lock.
by pbos@webrtc.org
· 11 years ago
ea5f28b
Adding call to Opus PLC
by tina.legrand@webrtc.org
· 11 years ago
7b0ab2a
Added logic for kSelectiveErrors to VCMJitterBuffer and corresponding unit tests.
by agalusza@google.com
· 11 years ago
b3ada15
Ref-counted rewrite of ChannelManager.
by pbos@webrtc.org
· 11 years ago
d7b06ec
Code formatting on files touched in r4447.
by pbos@webrtc.org
· 11 years ago
280c0b9
Added configuration of max delay to ACM and NetEq
by pwestin@webrtc.org
· 11 years ago
cda8ac1
Added Decoding with errors API to video_coding.h and removed unused DecodeError enum.
by agalusza@google.com
· 11 years ago
3166042
Add turaj@webrtc.org to NetEq owners.
by turaj@webrtc.org
· 11 years ago
f3bae63
Disabled flaky HardwareTest.BuiltInWasapiAECWorksForAudioWindowsCoreAudioLayer.
by phoglund@webrtc.org
· 11 years ago
44634a6
Disabled SsrcPropagatesCorrectly on Linux.
by phoglund@webrtc.org
· 11 years ago
71ffa0c
Better error treatment in NetEqImpl::InsertPacketInternal()
by minyue@webrtc.org
· 11 years ago
2d3071f
removed NetEq::EnableDtmf()
by minyue@webrtc.org
· 11 years ago
ea7b33e
* Update libjingle to 50389769.
by wu@webrtc.org
· 11 years ago
5978712
Invert dependency between webrtc_utility and media_file targets to reflect reality.
by fischman@webrtc.org
· 11 years ago
3ddbca9
Updated WebRTC version number to 3.38
by elham@webrtc.org
· 11 years ago
3f45c2e
Switch C++-style C headers with their C equivalents.
by pbos@webrtc.org
· 11 years ago
043f6a8
Fix implicit int->bool conversion in VideoSendStream::DeliverRtcp.
by pbos@webrtc.org
· 11 years ago
78ab511
Use RtpHeaderParser in VideoCall implementation.
by pbos@webrtc.org
· 11 years ago
ce85109
Glue code and tests for NACK in new VideoEngine API.
by pbos@webrtc.org
· 11 years ago
3a74d40
Fix send times in video_full_stack.
by pbos@webrtc.org
· 11 years ago
8704595
Add back is.FrameProvider() call lost in r4194.
by pbos@webrtc.org
· 11 years ago
146fd3c
Remove redundant conditions key.
by andrew@webrtc.org
· 11 years ago
4b8077b
Add one API for implementing Initial delay.
by turaj@webrtc.org
· 11 years ago
acb00f5
Adds all unittests to android NDK-APK framework.
by henrike@webrtc.org
· 11 years ago
478d711
Add some virtual and OVERRIDEs in webrtc/common_audio/
by pbos@webrtc.org
· 11 years ago
24add92
Fix some chromium-style warnings in webrtc/modules/audio_processing/
by pbos@webrtc.org
· 11 years ago
c7df0aa
Fix crash in DesktopRegion::Intersect().
by sergeyu@chromium.org
· 11 years ago
7affcd2
Fix some chromium-style warnings in webrtc/system_wrappers/
by pbos@webrtc.org
· 11 years ago
7b2147f
Removed lines preventing simultaneous kHardNack and decoding with errors. Also made changes recommended by gcl lint (with the exception of changing non-const references to pointers).
by agalusza@google.com
· 11 years ago
aa79e6e
Unbreak clang/android build of webrtc.
by fischman@webrtc.org
· 11 years ago
cb9a72b
Adding possibility to use encoding time when trigger underuse for frame based overuse detection.
by mflodman@webrtc.org
· 11 years ago
5ce8723
Merge r4374 from stable to trunk.
by xians@webrtc.org
· 11 years ago
0e6fa8c
Merge r4394 from stable to trunk.
by xians@webrtc.org
· 11 years ago
44f1239
Merge r4326 from stable to trunk.
by xians@webrtc.org
· 11 years ago
60bf21e
Handel zero correlation if at the same time distortion is also zero.
by turaj@webrtc.org
· 11 years ago
dd1b19d
Add some virtual and OVERRIDEs in webrtc/modules/audio_coding/
by pbos@webrtc.org
· 11 years ago
10b3664
Fix some chromium-style warnings in webrtc/modules/desktop_capture/
by pbos@webrtc.org
· 11 years ago
54042b9
Fix some chromium-style warnings in webrtc/modules/pacing/
by pbos@webrtc.org
· 11 years ago
9d71e28
Fix some chromium-style warnings in webrtc/modules/rtp_rtcp/
by pbos@webrtc.org
· 11 years ago
988a5b3
Fix some chromium-style warnings in webrtc/modules/remote_bitrate_estimator/
by pbos@webrtc.org
· 11 years ago
51cd3c7
Fix some chromium-style warnings in webrtc/modules/bitrate_controller/
by pbos@webrtc.org
· 11 years ago
87ae00a
Added libjingle_peerconnection_java_unittest to buildbot_tests.py
by phoglund@webrtc.org
· 11 years ago
3ed68d4
Move internal aec_core defines out of header.
by andrew@webrtc.org
· 11 years ago
b3b9e5a
Add svn:ignore properties for all spuriously-removed dirs on Linux64 Release (internal).
by fischman@webrtc.org
· 11 years ago
e39e35f
Correcting Turaj's email.
by turaj@webrtc.org
· 11 years ago
6907050
Fix some chromium-style warnings in webrtc/modules/video_coding/
by pbos@webrtc.org
· 11 years ago
77c6d71
Fix some chromium-style warnings in webrtc/test/
by pbos@webrtc.org
· 11 years ago
08a3b0d
Fix some chromium-style warnings in webrtc/tools/
by pbos@webrtc.org
· 11 years ago
d8daa60
Fix some chromium-style warnings in webrtc/modules/audio_device/
by pbos@webrtc.org
· 11 years ago
99199e5
Sets up framework for decoding with errors: collects frame sizes (in number of packets) in JB and passes this information to VCMSessionInfo with rtt_ms as FrameData.
by agalusza@google.com
· 11 years ago
2dd408e
PeerConnectionTest.java: make the test work for the bots' v4l2loopback.
by fischman@webrtc.org
· 11 years ago
09121dc
Land http://webrtc-codereview.appspot.com/1632005/
by niklas.enbom@webrtc.org
· 11 years ago
45e69ce
Updated WebRTC version to 3.37 TBR=tnakamura@webrtc.org
by elham@webrtc.org
· 11 years ago
3081f6d
Improved error messages when binaries are missing. Also stderr = stdout now.
by phoglund@webrtc.org
· 11 years ago
d91e5ee
To fix a bug in InverseFFTAndWindow() function in AECM.
by kma@webrtc.org
· 11 years ago
33f81b1
Updated WebRtcNsx_PrepareSpectrumNeon() in accordance with the new real FFT interface in APM. For reference, you can check https://webrtc-codereview.appspot.com/1830004/diff/92001/webrtc/modules/audio_processing/ns/nsx_core.c, line 594 "static void PrepareSpectrumC()".
by kma@webrtc.org
· 11 years ago
d69e2f4
Access receiving_ under receive_cs critical section
by braveyao@webrtc.org
· 11 years ago
0496413
Don't set clang_use_chrome_plugins in common.gypi
by sergeyu@chromium.org
· 11 years ago
f09b0e3
Fixes resources and data path in modules_unittests.isolate.
by henrike@webrtc.org
· 11 years ago
05f2131
Downstream latest Chromium SincResampler changes.
by andrew@webrtc.org
· 11 years ago
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